1. Field of the Invention
This invention relates to a speech encoding method and apparatus in which an input speech signal is split on the time axis and encoded from one pre-set encoding unit to another. The invention also relates to an associated speech decoding method and apparatus.
2. Description of the Related Art
Up to now, there are known a variety of encoding methods for performing signal compression by exploiting statistic properties in the time domain and frequency domain of audio signals, inclusive of speech and acoustic signals, and human psychoacoustic properties. These encoding methods are roughly classified into encoding in the time domain, encoding in the frequency domain and analysis-synthesis encoding.
Among the techniques for high-efficiency encoding of speech signals, there are known sinusoidal analysis encoding, such as harmonic encoding or multi-band excitation (MBE) encoding, sub-band coding (SBC), linear predictive coding (LPC), discrete cosine transform (DCT), modified DCT (MDCT) and fast Fourier transform (FFT).
However, in the conventional harmonic coding for LPC residuals, the V/UV decision on the speech signals is a one-of-two type decision between V and UV, such that the reproduced sound for the voiced speech portion tends to be a buzzing sound.
For preventing this from occurring, the decoder side adds noise to the voiced speech portion in outputting the playback sound. However, with this method, the degree of addition of the noise is difficult to set because addition of excessive noise results in noisy playback speech, whereas addition of insufficient noise results in the buzzing playback speech.